1. Technical Field
The present invention relates generally to establishing communication between two modems and, more particularly, is directed to an apparatus and method for propagating an answer tone between modems in communication over a data network.
2. Description of the Prior Art
Land based telephone systems have been in existence for many years. These telephone networks use twisted pair copper telephone wire to connect telephones and other equipment to a centralized Public Switched Telephone Network (PSTN). The connection to the PSTN is normally made through a Central Office (CO). Until recently, traditional telephony functions such as voice conversations had been the predominant type of traffic carried by the PSTN. With the advent of the Internet and the World Wide Web, the twisted pair wires in existing telephone networks are carrying increasing amounts of digitized data.
Full duplex voice communications are carried either by 4-wire circuits or by 2-wire circuits. The 2-wire circuits are usually used in a local loop portion of the telephone system. The local loop is the connection between the telephone subscriber and the telephone central office. Four wire circuits have traditionally been used for long-haul communications, e.g. from one central office to another. Four-wire circuits typically comprise a 2-wire send path and a 2-wire receive path.
Hybrid circuits are used to convert the 2-wire local loop into a 4-wire circuit. Since these hybrid circuits are not perfect, they may reflect a portion of the signal received on the receive path of a 4-wire circuit back into the send path. This results in an echo. In order to combat the echo problem, echo cancelers and echo suppressors have been introduced into the PSTN signal path. Echo suppressors impose a time-varying attenuation of the signal received on the receive path of a long-haul connection. Echo cancelers anticipate the echo that may be reflected into the receive path (from the “send path” at the remote end of the connection) and attempt to subtract only the anticipated echo rather than attenuating the entire signal.
Facsimile (fax) machines and analog modems communicate digital data over analog telephone lines. Fax machines and modems are designed to convey outgoing digital data streams as modulated analog signals. Accordingly, fax machines and modems demodulate incoming analog signals and convert them into a digital data stream. Echo suppressors that may be present in the telephone signal path may cause problems for fax machines and analog modems. Because some types of modems employ their own echo cancelers, echo cancelers comprising the telephone infrastructure may need to be disabled.
In an effort to standardize methods for disabling echo cancelers and suppressors, the Telecommunication Standardization Sector of the International Telecommunications Union (ITU-T) has defined a “G-series” of design recommendations. These design recommendations, which include G.164, G.165 and G.168, define how echo suppressors and cancelers may be disabled through the use of an answer tone. Hence, an echo suppressor or canceler that is designed according to these recommendations will disable its function when it perceives a particular answer tone.
Modem and fax design specifications have also been promulgated by the industry. These specifications specify the use of an answer tone by a modem or fax machine when an echo suppressor or canceler needs to be disabled. These modem and fax design specifications roughly correspond to the “G-series” recommendations that steer the design of echo suppressors and echo cancelers. Three significant modem specifications are the V.25 specification, the V.32 specification and the V.8 specification.
The G.164 Recommendation is directed towards echo suppressors. Fax machines are half-duplex in nature. Hence, an echo suppressor that attenuates the receive path of a 4-wire circuit may adversely affect the operation of a fax machine. Fax machines conforming to ITU-T Recommendation T.30 use a simple 2100 Hz answer tone, as specified in the G.164 Recommendation, to disable echo suppressors. This variant of an answer tone does not disable echo cancelers. Fax machines actually rely on echo cancelers to eliminate echo artifacts from the receive path which might otherwise be misinterpreted as the beginning of a new transmission from a remote fax machine.
The G.165 and G.168 recommendations specify a means for disabling echo cancelers present in the telephone signal path servicing a modem connection. This allows modems to use their own internal echo cancellation without interference from the echo cancelers present in the telephone signal path connecting two modems. In accordance with G.165 and G.168, echo cancelers will disable themselves when they detect a 2100 Hz tone that periodically exhibits 180° phase reversals. The phase reversals typically occur every 450 milliseconds (ms). The V.8 modem specification defines yet another variant of an answer tone. The 2100 Hz tone with phase reversals, as described in the G.165 recommendation, is amplitude modulated by a low frequency carrier of 15 Hz.
Through evolution of modem technology, a wide variety of answer tones have been defined and are now in use. The type of answer tone used by a particular type of fax machine or modem may depend on the type of echo suppression or echo cancellation that may need to be disabled or maintained as two modem devices communicate with each other over a phone connection.
Modern telephone infrastructures are based primarily on digitized communications systems. Typically, a central office services a subscriber's facility with an analog telephone connection. In the central office, the analog signaling used to communicate with a telephone, modem or fax machine at a subscribers facility is typically converted to a stream of inbound and outbound data. The conversion is typically accomplished by a circuit called coder/decoder (codec). The digitized data can then be conveyed to a public switched telephone network (PSTN). The PSTN is formed by a series of interconnected time division multiplexed, constant rate data networks. The PSTN is a controlled latency digital communications network that can be used to establish “virtual circuits” from one subscriber to another. From a subscribers perspective, voice communications carried by the PSTN are clear and instantaneous.
Controlled latency networks are expensive to install and operate. Hence, it becomes advantageous to telephone companies and subscribers alike if viable alternatives to the PSTN could be used. Many telephone companies have started using general purpose data networks to carry voice communications on an experimental basis. One class of voice service supported by general purpose data networks is known as Voice over Internet Protocol (VoIP).
VoIP service digitizes analog telephony and uses a packet data network to establish a voice channel with another subscriber. Even though VoIP attempts to control quality, this class of service is characterized by significantly garbled and delayed voice quality due to the fact that a packet data network cannot control network jitter or latency as well as the PSTN. Even more disabling to VoIP quality is the fact that a packet network can loose packets altogether; a phenomenon not exhibited in PSTN communications.
In order to establish a VoIP connection, the telephone company central office must still digitize the analog telephone signals that it receives from the subscriber's facility. However, in contrast to standard telephone service where the digitized telephony is conveyed to the controlled latency PSTN, the digitized telephony signal is directed to a packet network. This is typically accomplished by a packet network access gateway that digitized the subscriber's signal and directs the digitized signal to a packet network. In one common alternative structure, the access gateway may receive digitized data from a subscriber interface unit (SIU) that may intelligently select VoIP when it is prudent or allowable to do so. The SIU typically comprises a codec and the requisite control circuitry to convey digitized telephony to either the PSTN network or to the access gateway.
Another problem associated with VoIP service pertains to the use of modems or fax machines over a voice channel that is carried by a digital communications channel established through a packet network. A VoIP connection typically comprises an analog phone in communication with another analog phone over a data network. The connection is typically established when a calling phone transmits an analog signal to a calling-side gateway. The analog signal is received by the calling-side gateway and is digitized using a codec. The digitized signal may then be transmitted over the data network from the calling-side gateway to an answering-side gateway. The answering-side gateway typically receives the digitized telephony data and converts it to an analog signal using another codec. This analog signal may then be directed to the answering telephone instrument.
When a fax machine or modem is attached to an analog telephone line serviced by an access gateway, the modulated analog signal it generates must be converted to a digital data stream by a codec. This digital data stream can then be conveyed to an answering-side gateway using a data network. This is conspicuously inefficient in light of the fact that the fax machine or modem originally created the modulated analog signal from a digital data stream. It would be more efficient to convey the original data stream over the data network in lieu of a digitized rendition of the modulated analog signal.
A more efficient way of operating fax machines or modems in conjunction with an access gateway is a relatively new class of service called Modem over Internet Protocol (MoIP). In MoIP, an access gateway further comprises a modem that intercepts the modulated analog signal generated by a fax machine or a modem connected to an analog telephone interface comprising the gateway. A calling modem receives digital data from a computer and modulates an analog signal according to the digital data. The modulated analog signal is transmitted to a calling-side gateway where it is demodulated by the modem comprising the gateway and converted back into a digital data stream. The digital data stream is then conveyed across the data network to an answering-side gateway. The answering-side gateway receives the digital data stream from the data network. A modem comprising the answering-side gateway converts the received digital data stream into a modulated analog signal. The modulated analog signal may then be directed to an answering modem.
When a modem first initiates a call through a calling-side gateway, the calling-side gateway presumes the call originated from a telephone. In response, a voice channel is established between the calling-side gateway and an answering-side gateway by creating a VoIP connection. The VoIP connection is completed when the answering-side gateway propagates the telephone call to an answering modem.
When the answering modem answers the VoIP call, it responds by transmitting an answer tone. When the answering-side gateway detects the answer tone from the answering modem, it realizes that a modem or fax session is about the begin using the VoIP connection. Responding to the answer tone, the answering-side gateway tears down the VoIP connection and establishes a MoIP connection. The answering-side gateway typically prevents the answer tone from reaching the calling-side gateway through the VoIP connection.
In furtherance of establishing the MoIP connection, the answering-side gateway signals the calling-side gateway that it has detected an answer tone. Once the calling-side gateway receives this signal, it regenerates the answer tone and conveys the regenerated tone back to the calling modem. Typically, this signal indicates not only that the answering-side gateway received an answer tone; it also indicates the type of answer tone received.
One problem with MoIP is that modems and Fax machines that are designed to be used over the PSTN network often fail to connect when attempting to communicate over a data network. This is sometimes due to the latency associated with regenerating the answer tone back to the calling modem from the calling-side gateway. A significant amount of time may be lost as an answering-side gateway attempts to identify the type of answer tone received prior to signaling the calling-side gateway.